RFC5219
Network Working Group R. Finlayson Request for Comments: 5219 Live Networks, Inc. Obsoletes: 3119 February 2008 Category: Standard Track
A More Loss-Tolerant RTP Payload Format for MP3 Audio
Status of This Memo
This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards" (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited.
Abstract
This document describes an RTP (Real-Time Protocol) payload format for transporting MPEG (Moving Picture Experts Group) 1 or 2, layer III audio (commonly known as "MP3"). This format is an alternative to that described in RFC 2250, and performs better if there is packet loss. This document obsoletes RFC 3119, correcting typographical errors in the "SDP usage" section and pseudo-code appendices.
A.1. Converting a Sequence of "MP3 Frames"
A.2. Converting a Sequence of "ADU Frames"
B.2. Deinterleaving a Sequence of (Interleaved) "ADU Frames" ...19
Contents
Introduction
While the RTP payload format defined in RFC 2250 [1] is generally applicable to all forms of MPEG audio or video, it is sub-optimal for MPEG-1 or 2, layer III audio (commonly known as "MP3"). The reason for this is that an MP3 frame is not a true "Application Data Unit" -- it contains a back-pointer to data in earlier frames, and so cannot be decoded independently of these earlier frames. Because RFC 2250 defines that packet boundaries coincide with frame boundaries, it handles packet loss inefficiently when carrying MP3 data. The loss of an MP3 frame will render some data in previous (or future) frames useless, even if they are received without loss.
In this document, we define an alternative RTP payload format for MP3 audio. This format uses a data-preserving rearrangement of the original MPEG frames, so that packet boundaries now coincide with true MP3 "Application Data Units", which can also (optionally) be rearranged in an interleaving pattern. This new format is therefore more data efficient than RFC 2250 in the face of packet loss.
Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [2].
The Structure of MP3 Frames
In this section we give a brief overview of the structure of an MP3 frame. (For a more detailed description, see the MPEG-1 audio [3] and MPEG-2 audio [4] specifications.)
Each MPEG audio frame begins with a 4-byte header. Information defined by this header includes:
- Whether the audio is MPEG-1 or MPEG-2. - Whether the audio is layer I, II, or III. (The remainder of this
document assumes layer III, i.e., "MP3" frames.)
- Whether the audio is mono or stereo. - Whether or not there is a 2-byte CRC field following the header. - (indirectly) The size of the frame.
The following structures appear after the header:
- (optionally) A 2-byte Cyclic Redundancy Check (CRC) field - A "side info" structure. This has the following length:
- 32 bytes for MPEG-1 stereo - 17 bytes for MPEG-1 mono, or for MPEG-2 stereo - 9 bytes for MPEG-2 mono
- Encoded audio data, plus optional ancillary data (filling out the
rest of the frame)
For the purpose of this document, the "side info" structure is the most important, because it defines the location and size of the "Application Data Unit" (ADU) that an MP3 decoder will process. In particular, the "side info" structure defines:
- "main_data_begin": This is a back-pointer (in bytes) to the start
of the ADU. The back-pointer is counted from the beginning of the frame, and counts only encoded audio data and any ancillary data (i.e., ignoring any header, CRC, or "side info" fields).
An MP3 decoder processes each ADU independently. The ADUs will generally vary in length, but their average length will, of course,
be that of the of the MP3 frames (minus the length of the header, CRC, and "side info" fields). (In MPEG literature, this ADU is sometimes referred to as a "bit reservoir".)
A New Payload Format
As noted in [5], a payload format should be designed so that packet boundaries coincide with "codec frame boundaries" -- i.e., with ADUs. In the RFC 2250 payload format for MPEG audio [1], each RTP packet payload contains MP3 frames. In this new payload format for MP3 audio, however, each RTP packet payload contains "ADU frames", each preceded by an "ADU descriptor".
ADU Frames
An "ADU frame" is defined as:
- The 4-byte MPEG header (the same as the original MP3 frame,
except that the first 11 bits are (optionally) replaced by an
"Interleaving Sequence Number", as described in Section 7
below)
- The optional 2-byte CRC field (the same as the original MP3
frame)
- The "side info" structure (the same as the original MP3 frame)
- The complete sequence of encoded audio data (and any ancillary
data) for the ADU (i.e., running from the start of this MP3
frame's "main_data_begin" back-pointer, up to the start of the
next MP3 frame's back-pointer)
Note that there is a one-to-one mapping between MP3 frames and ADU frames. Because MP3 frames are self-describing, with the bitrate (and sampling frequency) encoded within the 4-byte MPEG header, the same is true for ADU frames. Therefore, as with MP3 streams, the bitrate can change within a stream and may be used for congestion control.
ADU Descriptors
Within each RTP packet payload, each "ADU frame" is preceded by a 1- or 2-byte "ADU descriptor", which gives the size of the ADU and indicates whether or not this packet's data is a continuation of the previous packet's data. (This occurs only when a single "ADU descriptor" + "ADU frame" is too large to fit within an RTP packet.)
An ADU descriptor consists of the following fields:
- "C": Continuation flag (1 bit): 1, if the data following the ADU
descriptor is a continuation of an ADU frame that was too
large to fit within a single RTP packet; 0 otherwise.
- "T": Descriptor Type flag (1 bit):
0 if this is a 1-byte ADU descriptor;
1 if this is a 2-byte ADU descriptor.
- "ADU size" (6 or 14 bits): The size (in bytes) of the ADU frame
that will follow this ADU descriptor (i.e., NOT including the
size of the descriptor itself). A 2-byte ADU descriptor
(with a 14-bit "ADU size" field) is used for ADU frame sizes
of 64 bytes or more. For smaller ADU frame sizes, senders
MAY alternatively use a 1-byte ADU descriptor (with a 6-bit
"ADU size" field). Receivers MUST be able to accept an ADU
descriptor of either size.
Thus, a 1-byte ADU descriptor is formatted as follows:
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|C|0| ADU size |
+-+-+-+-+-+-+-+-+
and a 2-byte ADU descriptor is formatted as follows:
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|C|1| ADU size (14 bits) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Packing Rules
Each RTP packet payload begins with an "ADU descriptor", followed by "ADU frame" data. Normally, this "ADU descriptor" + "ADU frame" will fit completely within the RTP packet. In this case, more than one successive "ADU descriptor" + "ADU frame" MAY be packed into a single RTP packet, provided that they all fit completely.
If, however, a single "ADU descriptor" + "ADU frame" is too large to fit within an RTP packet, then the "ADU frame" is split across two or more successive RTP packets. Each such packet begins with an ADU descriptor. The first packet's descriptor has a "C" (continuation) flag of 0; the following packets' descriptors each have a "C" flag of 1. Each descriptor, in this case, has the same "ADU size" value: the
size of the entire "ADU frame" (not just the portion that will fit within a single RTP packet). Each such packet (even the last one) contains only one "ADU descriptor".
RTP Header Fields
Payload Type: The (static) payload type 14 that was defined for
MPEG audio [6] MUST NOT be used. Instead, a different, dynamic payload type MUST be used -- i.e., one within the range [96..127].
M bit: This payload format defines no use for this bit. Senders
SHOULD set this bit to zero in each outgoing packet.
Timestamp: This is a 32-bit, 90 kHz timestamp, representing the
presentation time of the first ADU packed within the packet.
Handling Received Data
Note that no information is lost by converting a sequence of MP3 frames to a corresponding sequence of "ADU frames", so a receiving RTP implementation can either feed the ADU frames directly to an appropriately modified MP3 decoder, or convert them back into a sequence of MP3 frames, as described in Appendix A.2 below.
Handling Multiple MPEG Audio Layers
The RTP payload format described here is intended only for MPEG-1 or 2, layer III audio ("MP3"). In contrast, layer I and layer II frames are self-contained, without a back-pointer to earlier frames. However, it is possible (although unusual) for a sequence of audio frames to consist of a mixture of layer III frames, and layer I or II frames. When such a sequence is transmitted, only layer III frames are converted to ADUs; layer I or II frames are sent 'as is' (except for the prepending of an "ADU descriptor"). Similarly, the receiver of a sequence of frames -- using this payload format -- leaves layer I and II frames untouched (after removing the prepended "ADU descriptor"), but converts layer III frames from "ADU frames" to regular MP3 frames. (Recall that each frame's layer is identified from its 4-byte MPEG header.)
If you are transmitting a stream consisting *only* of layer I or layer II frames (i.e., without any MP3 data), then there is no benefit to using this payload format, *unless* you are using the interleaving mechanism described in Section 7 below.
Frame Packetizing and Depacketizing
The transmission of a sequence of MP3 frames takes the following steps:
MP3 frames
-1-> ADU frames
-2-> interleaved ADU frames
-3-> RTP packets
Step 1 is the conversion of a sequence of MP3 frames to a corresponding sequence of ADU frames, and takes place as described in Sections 3 and 4.1 above. (Note also the pseudo-code in Appendix A.1.)
Step 2 is the reordering of the sequence of ADU frames in an (optional) interleaving pattern, prior to packetization, as described in section 7 below. (Note also the pseudo-code in Appendix B.1.) Interleaving helps reduce the effect of packet loss by distributing consecutive ADU frames over non-consecutive packets. (Note that because of the back-pointer in MP3 frames, interleaving can be applied -- in general -- only to ADU frames. Thus, interleaving was not possible for RFC 2250.)
Step 3 is the packetizing of a sequence of (interleaved) ADU frames into RTP packets -- as described in section 4.3 above. Each packet's RTP timestamp is the presentation time of the first ADU that is packed within it. Note that if interleaving was done in step 2, the RTP timestamps on outgoing packets will not necessarily be monotonically nondecreasing.
Similarly, a sequence of received RTP packets is handled as follows:
RTP packets
-4-> RTP packets ordered by RTP sequence number
-5-> interleaved ADU frames
-6-> ADU frames
-7-> MP3 frames
Step 4 is the usual sorting of incoming RTP packets using the RTP sequence number.
Step 5 is the depacketizing of ADU frames from RTP packets -- i.e., the reverse of step 3. As part of this process, a receiver uses the "C" (continuation) flag in the ADU descriptor to notice when an ADU frame is split over more than one packet (and to discard the ADU frame entirely if one of these packets is lost).
Step 6 is the rearranging of the sequence of ADU frames back to its original order (except for ADU frames missing due to packet loss), as described in Section 7 below. (Note also the pseudo-code in Appendix B.2.)
Step 7 is the conversion of the sequence of ADU frames into a corresponding sequence of MP3 frames -- i.e., the reverse of step 1. (Note also the pseudo-code in Appendix A.2.) With an appropriately modified MP3 decoder, an implementation may omit this step; instead, it could feed ADU frames directly to the (modified) MP3 decoder.
ADU Frame Interleaving
In MPEG audio frames (MPEG-1 or 2; all layers), the high-order 11 bits of the 4-byte MPEG header ('syncword') are always all-one (i.e., 0xFFE). When reordering a sequence of ADU frames for transmission, we reuse these 11 bits as an "Interleaving Sequence Number" (ISN). (Upon reception, they are replaced with 0xFFE once again.)
The structure of the ISN is (a,b), where:
- a == bits 0-7: 8-bit Interleave Index (within Cycle)
- b == bits 8-10: 3-bit Interleave Cycle Count
That is, the 4-byte MPEG header is reused as follows:
0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |Interleave Idx |CycCt| The rest of the original MPEG header | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Example: Consider the following interleave cycle (of size 8):
1,3,5,7,0,2,4,6
(This particular pattern has the property that any loss of up to four consecutive ADUs in the interleaved stream will lead to a deinterleaved stream with no gaps greater than one.) This produces the following sequence of ISNs:
(1,0) (3,0) (5,0) (7,0) (0,0) (2,0) (4,0) (6,0) (1,1) (3,1) (5,1) etc.
So, in this example, a sequence of ADU frames
f0 f1 f2 f3 f4 f5 f6 f7 f8 f9 (etc.)
would get reordered, in step 2, into:
(1,0)f1 (3,0)f3 (5,0)f5 (7,0)f7 (0,0)f0 (2,0)f2 (4,0)f4 (6,0)f6 (1,1)f9 (3,1)f11 (5,1)f13 (etc.)
and the reverse reordering (along with replacement of the 0xFFE) would occur upon reception.
The reason for breaking the ISN into "Interleave Cycle Count" and "Interleave Index" (rather than just treating it as a single 11-bit counter) is to give receivers a way of knowing when an ADU frame should be 'released' to the ADU->MP3 conversion process (step 7 above), rather than waiting for more interleaved ADU frames to arrive. For instance, in the example above, when the receiver sees a frame with ISN (<something>,1), it knows that it can release all previously seen frames with ISN (<something>,0), even if some other (<something>,0) frames remain missing due to packet loss. An 8-bit Interleave Index allows interleave cycles of size up to 256.
The choice of an interleaving order can be made independently of RTP packetization. Thus, a simple implementation could choose an interleaving order first, reorder the ADU frames accordingly (step 2), then simply pack them sequentially into RTP packets (step 3). However, the size of ADU frames -- and thus the number of ADU frames that will fit in each RTP packet -- will typically vary in size, so a more optimal implementation would combine steps 2 and 3, by choosing an interleaving order that better reflected the number of ADU frames packed within each RTP packet.
Each receiving implementation of this payload format MUST recognize the ISN and be able to perform deinterleaving of incoming ADU frames (step 6). However, a sending implementation of this payload format MAY choose not to perform interleaving -- i.e., by omitting step 2. In this case, the high-order 11 bits in each 4-byte MPEG header would remain at 0xFFE. Receiving implementations would thus see a sequence of identical ISNs (all 0xFFE). They would handle this in the same way as if the Interleave Cycle Count changed with each ADU frame, by simply releasing the sequence of incoming ADU frames sequentially to the ADU->MP3 conversion process (step 7), without reordering. (Note also the pseudo-code in Appendix B.2.)
IANA Considerations
Media type name: audio
Media subtype: mpa-robust
Required parameters: none
Optional parameters: none
Encoding considerations:
This type is defined only for transfer via RTP, as specified in
RFC 5219.
Security considerations:
See the "Security Considerations" section of RFC 5219.
Interoperability considerations:
This encoding is incompatible with both the "audio/mpa" and
"audio/mpeg" media types.
Published specification:
The ISO/IEC MPEG-1 [3] and MPEG-2 [4] audio specifications, and
RFC 5219.
Applications that use this media type:
Audio streaming tools (transmitting and receiving)
Additional information: none
Person & email address to contact for further information:
Ross Finlayson
[email protected]
Intended usage: COMMON
Author/Change controller:
Author: Ross Finlayson
Change controller: IETF AVT Working Group
SDP Usage
When conveying information by SDP [7], the encoding name SHALL be "mpa-robust" (the same as the media subtype). An example of the media representation in SDP is:
m=audio 49000 RTP/AVP 121
a=rtpmap:121 mpa-robust/90000
Note that the RTP timestamp frequency MUST be 90000.
10. Security Considerations
If a session using this payload format is being encrypted, and interleaving is being used, then the sender SHOULD ensure that any change of encryption key coincides with a start of a new interleave cycle. Apart from this, the security considerations for this payload format are identical to those noted for RFC 2250 [1].
11. Acknowledgements
The suggestion of adding an interleaving option (using the first bits of the MPEG 'syncword' -- which would otherwise be all-ones -- as an interleaving index) is due to Dave Singer and Stefan Gewinner. In addition, Dave Singer provided valuable feedback that helped clarify
and improve the description of this payload format. Feedback from Chris Sloan led to the addition of an "ADU descriptor" preceding each ADU frame in the RTP packet.
12. Normative References
[1] Hoffman, D., Fernando, G., Goyal, V., and M. Civanlar, "RTP
Payload Format for MPEG1/MPEG2 Video", RFC 2250, January 1998.
[2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[3] ISO/IEC International Standard 11172-3; "Coding of moving
pictures and associated audio for digital storage media up to about 1,5 Mbits/s -- Part 3: Audio", 1993.
[4] ISO/IEC International Standard 13818-3; "Generic coding of moving
pictures and associated audio information -- Part 3: Audio", 1998.
[5] Handley, M. and C. Perkins, "Guidelines for Writers of RTP
Payload Format Specifications", BCP 36, RFC 2736, December 1999.
[6] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
[7] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
Appendix A. Translating between "MP3 Frames" and "ADU Frames"
The following 'pseudo code' describes how a sender using this payload format can translate a sequence of regular "MP3 Frames" to "ADU Frames", and how a receiver can perform the reverse translation: from "ADU Frames" to "MP3 Frames".
We first define the following abstract data structures:
- "Segment": A record that represents either a "MP3 Frame" or an
"ADU Frame". It consists of the following fields:
- "header": the 4-byte MPEG header
- "headerSize": a constant (== 4)
- "sideInfo": the 'side info' structure, *including* the optional
2-byte CRC field, if present
- "sideInfoSize": the size (in bytes) of the above structure
- "frameData": the remaining data in this frame
- "frameDataSize": the size (in bytes) of the above data
- "backpointer": the value (expressed in bytes) of the
backpointer for this frame
- "aduDataSize": the size (in bytes) of the ADU associated with
this frame. (If the frame is already an "ADU Frame", then
aduDataSize == frameDataSize)
- "mp3FrameSize": the total size (in bytes) that this frame would
have if it were a regular "MP3 Frame". (If it is already a
"MP3 Frame", then mp3FrameSize == headerSize + sideInfoSize +
frameDataSize) Note that this size can be derived completely
from "header".
- "SegmentQueue": A FIFO queue of "Segments", with operations
- void enqueue(Segment)
- Segment dequeue()
- Boolean isEmpty()
- Segment head()
- Segment tail()
- Segment previous(Segment): returns the segment prior to a
given one
- Segment next(Segment): returns the segment after a given one
- unsigned totalDataSize(): returns the sum of the
"frameDataSize" fields of each entry in the queue
A.1. Converting a Sequence of "MP3 Frames" to a Sequence of
"ADU Frames"
SegmentQueue pendingMP3Frames; // initially empty while (1) {
// Enqueue new MP3 Frames, until we have enough data to
// generate the ADU for a frame:
do {
int totalDataSizeBefore
= pendingMP3Frames.totalDataSize();
Segment newFrame = 'the next MP3 Frame';
pendingMP3Frames.enqueue(newFrame);
int totalDataSizeAfter
= pendingMP3Frames.totalDataSize();
} while (totalDataSizeBefore < newFrame.backpointer ||
totalDataSizeAfter < newFrame.aduDataSize);
// We now have enough data to generate the ADU for the most
// recently enqueued frame (i.e., the tail of the queue).
// (The earlier frames in the queue -- if any -- must be
// discarded, as we don't have enough data to generate
// their ADUs.)
Segment tailFrame = pendingMP3Frames.tail();
// Output the header and side info:
output(tailFrame.header);
output(tailFrame.sideInfo);
// Go back to the frame that contains the start of our
// ADU data:
int offset = 0;
Segment curFrame = tailFrame;
int prevBytes = tailFrame.backpointer;
while (prevBytes > 0) {
curFrame = pendingMP3Frames.previous(curFrame);
int dataHere = curFrame.frameDataSize;
if (dataHere < prevBytes) {
prevBytes -= dataHere;
} else {
offset = dataHere - prevBytes;
break;
}
}
// Dequeue any frames that we no longer need:
while (pendingMP3Frames.head() != curFrame) {
pendingMP3Frames.dequeue();
}
// Output, from the remaining frames, the ADU data that
// we want:
int bytesToUse = tailFrame.aduDataSize;
while (bytesToUse > 0) {
int dataHere = curFrame.frameDataSize - offset;
int bytesUsedHere
= dataHere < bytesToUse ? dataHere :
bytesToUse;
output("bytesUsedHere" bytes from
curFrame.frameData, starting from "offset");
bytesToUse -= bytesUsedHere;
offset = 0;
curFrame = pendingMP3Frames.next(curFrame);
}
}
A.2. Converting a Sequence of "ADU Frames" to a Sequence of
"MP3 Frames"
SegmentQueue pendingADUFrames; // initially empty while (1) {
while (needToGetAnADU()) {
Segment newADU = 'the next ADU Frame';
pendingADUFrames.enqueue(newADU);
insertDummyADUsIfNecessary();
}
generateFrameFromHeadADU();
}
Boolean needToGetAnADU() {
// Checks whether we need to enqueue one or more new ADUs
// before we have enough data to generate a frame for the
// head ADU.
Boolean needToEnqueue = True;
if (!pendingADUFrames.isEmpty()) {
Segment curADU = pendingADUFrames.head();
int endOfHeadFrame = curADU.mp3FrameSize
- curADU.headerSize - curADU.sideInfoSize;
int frameOffset = 0;
while (1) {
int endOfData = frameOffset
- curADU.backpointer +
curADU.aduDataSize;
if (endOfData >= endOfHeadFrame) {
// We have enough data to generate a
// frame.
needToEnqueue = False;
break;
}
frameOffset += curADU.mp3FrameSize
- curADU.headerSize
- curADU.sideInfoSize;
if (curADU == pendingADUFrames.tail()) break;
curADU = pendingADUFrames.next(curADU);
}
}
return needToEnqueue;
}
void generateFrameFromHeadADU() {
Segment curADU = pendingADUFrames.head();
// Output the header and side info:
output(curADU.header);
output(curADU.sideInfo);
// Begin by zeroing out the rest of the frame, in case the
// ADU data doesn't fill it in completely:
int endOfHeadFrame = curADU.mp3FrameSize
- curADU.headerSize - curADU.sideInfoSize;
output("endOfHeadFrame" zero bytes);
// Fill in the frame with appropriate ADU data from this and
// subsequent ADUs:
int frameOffset = 0;
int toOffset = 0;
while (toOffset < endOfHeadFrame) {
int startOfData = frameOffset - curADU.backpointer;
if (startOfData > endOfHeadFrame) {
break; // no more ADUs are needed
}
int endOfData = startOfData + curADU.aduDataSize;
if (endOfData > endOfHeadFrame) {
endOfData = endOfHeadFrame;
}
int fromOffset;
if (startOfData <= toOffset) {
fromOffset = toOffset - startOfData;
startOfData = toOffset;
if (endOfData < startOfData) {
endOfData = startOfData;
}
} else {
fromOffset = 0;
// leave some zero bytes beforehand:
toOffset = startOfData;
}
int bytesUsedHere = endOfData - startOfData;
output(starting at offset "toOffset", "bytesUsedHere"
bytes from "&curADU.frameData[fromOffset]");
toOffset += bytesUsedHere;
frameOffset += curADU.mp3FrameSize
- curADU.headerSize - curADU.sideInfoSize;
curADU = pendingADUFrames.next(curADU);
}
pendingADUFrames.dequeue();
}
void insertDummyADUsIfNecessary() {
// The tail segment (ADU) is assumed to have been recently
// enqueued. If its backpointer would overlap the data
// of the previous ADU, then we need to insert one or more
// empty, 'dummy' ADUs ahead of it. (This situation
// should occur only if an intermediate ADU was missing
// -- e.g., due to packet loss.)
while (1) {
Segment tailADU = pendingADUFrames.tail();
int prevADUend; // relative to the start
of the tail ADU
if (pendingADUFrames.head() != tailADU) {
// there is a previous ADU
Segment prevADU
= pendingADUFrames.previous(tailADU);
prevADUend
= prevADU.mp3FrameSize +
prevADU.backpointer
- prevADU.headerSize
- prevADU.sideInfoSize;
if (prevADU.aduDataSize > prevADUend) {
// this shouldn't happen if the
// previous ADU was well-formed
prevADUend = 0;
} else {
prevADUend -= prevADU.aduDataSize;
}
} else {
prevADUend = 0;
}
if (tailADU.backpointer > prevADUend) {
// Insert a 'dummy' ADU in front of the tail.
// This ADU can have the same "header" (and thus,
// "mp3FrameSize") as the tail ADU, but should
// have a "backpointer" of "prevADUend", and
// an "aduDataSize" of zero. The simplest
// way to do this is to copy the "sideInfo" from
// the tail ADU, replace the value of
// "main_data_begin" with "prevADUend", and set
// all of the "part2_3_length" fields to zero.
} else {
break; // no more dummy ADUs need to be
// inserted
}
}
}
Appendix B. Interleaving and Deinterleaving
The following 'pseudo code' describes how a sender can reorder a sequence of "ADU Frames" according to an interleaving pattern (step 2), and how a receiver can perform the reverse reordering (step 6).
B.1. Interleaving a Sequence of "ADU Frames"
We first define the following abstract data structures:
- "interleaveCycleSize": an integer in the range [1..256] --
"interleaveCycle": an array, of size "interleaveCycleSize", containing some permutation of the integers from the set [0 .. interleaveCycleSize-1] e.g., if "interleaveCycleSize" == 8, "interleaveCycle" might contain: 1,3,5,7,0,2,4,6
- "inverseInterleaveCycle": an array containing the inverse of the
permutation in "interleaveCycle" -- i.e., such that
interleaveCycle[inverseInterleaveCycle[i]] == i
- "ii": the current Interleave Index (initially 0) - "icc": the current Interleave Cycle Count (initially 0) - "aduFrameBuffer": an array, of size "interleaveCycleSize", of ADU
Frames that are awaiting packetization
while (1) {
int positionOfNextFrame = inverseInterleaveCycle[ii];
aduFrameBuffer[positionOfNextFrame] = the next ADU frame;
replace the high-order 11 bits of this frame's MPEG header
with (ii,icc);
// Note: Be sure to leave the remaining 21 bits
// as is
if (++ii == interleaveCycleSize) {
// We've finished this cycle, so pass all
// pending frames to the packetizing step
for (int i = 0; i < interleaveCycleSize; ++i) {
pass aduFrameBuffer[i] to the packetizing step;
}
ii = 0;
icc = (icc+1)%8;
}
}
B.2. Deinterleaving a Sequence of (Interleaved) "ADU Frames"
We first define the following abstract data structures:
- "ii": the Interleave Index from the current incoming ADU frame - "icc": the Interleave Cycle Count from the current incoming ADU
frame
- "iiLastSeen": the most recently seen Interleave Index (initially,
some integer *not* in the range [0..255])
- "iccLastSeen": the most recently seen Interleave Cycle Count
(initially, some integer *not* in the range [0..7])
- "aduFrameBuffer": an array, of size 256, of (pointers to) ADU
Frames that have just been depacketized (initially, all entries are NULL)
while (1) {
aduFrame = the next ADU frame from the depacketizing step;
(ii,icc) = "the high-order 11 bits of aduFrame's MPEG
header"; "the high-order 11 bits of aduFrame's MPEG
header" = 0xFFE;
// Note: Be sure to leave the remaining 21 bits
// as is
if (icc != iccLastSeen || ii == iiLastSeen) {
// We've started a new interleave cycle
// (or interleaving was not used). Release all
// pending ADU frames to the ADU->MP3 conversion
// step:
for (int i = 0; i < 256; ++i) {
if (aduFrameBuffer[i] != NULL) {
release aduFrameBuffer[i];
aduFrameBuffer[i] = NULL;
}
}
}
iiLastSeen = ii;
iccLastSeen = icc;
aduFrameBuffer[ii] = aduFrame;
}
Appendix C. Changes from RFC 3119
The primary change from RFC 3119 is to correct the encoding name in the "SDP usage" section. The correct encoding name is "mpa-robust". Also, the term "media type" replaces "mime type". Finally, some minor bug fixes and clarifications were made to the (non-normative) pseudo code in Appendix A and Appendix B.
Author's Address
Ross Finlayson, Live Networks, Inc. 650 Castro St., suite 120-196 Mountain View, CA 94041 USA
EMail: [email protected] URI: http://www.live555.com/
Full Copyright Statement
Copyright (C) The IETF Trust (2008).
This document is subject to the rights, licenses and restrictions contained in BCP 78, and except as set forth therein, the authors retain all their rights.
This document and the information contained herein are provided on an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST AND THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Intellectual Property
The IETF takes no position regarding the validity or scope of any Intellectual Property Rights or other rights that might be claimed to pertain to the implementation or use of the technology described in this document or the extent to which any license under such rights might or might not be available; nor does it represent that it has made any independent effort to identify any such rights. Information on the procedures with respect to rights in RFC documents can be found in BCP 78 and BCP 79.
Copies of IPR disclosures made to the IETF Secretariat and any assurances of licenses to be made available, or the result of an attempt made to obtain a general license or permission for the use of such proprietary rights by implementers or users of this specification can be obtained from the IETF on-line IPR repository at http://www.ietf.org/ipr.
The IETF invites any interested party to bring to its attention any copyrights, patents or patent applications, or other proprietary rights that may cover technology that may be required to implement this standard. Please address the information to the IETF at [email protected].
